Recent content by uzumaki

Hi guys, I have loaded SP10 in my ESS 6.0.1 and all LSP 6.0.1 sites. My main is on 6.0.1 SP1. Is it safe to failover to ESS while upgrading the main to SP10?

Hi Guys, Is there a way how to change the call appearance on the new 96x1 phones from extension number being shown by default to customized label that a user wants? I have 450 stations to change, can we possibly do it 46xx settings file?

Hi i have a customer who will migrate from CM4 to CM6. My customer is currently using the old Call Master IV and V. Some functionality like login, logout, aux button when you program on the call masters can have a GREEN LED status on the side once it is pressed and it will NOT go out. But on...

Hi Guys, I'm trying to design a solution that will migrate my customer from a traditional H323 stations to SIP. I have smgr, asm, cm and aes in place. I havent done the final BOM yet and its still on the drawing board. I'm not so sure if the call recording of verint (using TSAPI and DMCC via...

Hi Guys, I'm trying to design a solution that will migrate my customer from a traditional H323 stations to SIP. I have smgr, asm, cm and aes in place. I havent done the final BOM yet and its still on the drawing board. I'm not so sure if the call recording of verint (using TSAPI and DMCC via...

1. Add DS1 #### (Setup the parameters for line coding, framing mode, signaling mode, connect, Interworking Message, Interface companding, country protocol, Protocol, version, CRC and Splip detection 2. Create signaling and define the ds1 port 3. Create trunk and define the DS1 board port in step...

Sorry to ask, but usually how does this converse-on work. Is it similar to adjunct routing? Basically what I understand on adjunct routing, "your giving the full control on the 3rd party equipment to do call routing". What about this converse-on step?

My current setup is: CM5.2.1 <------> ASM 6.1 <------> MM5.2 I have station A, B, C, D (9620, 9640 and 4620) each station has a cov path 10 (after 2 ring go to voicemail hunt) If I call either B, C or D using station A when when go to cover and MM answers its always saying sorry cannot...

It worked. So the logic is if a bridged station wants to make a call, he needs to select an idle appearance and not chose the line appearance that is busy or has a new call?

I have station A B and C. Station B is bridged to station A. Station C calls Station A, and at the same time B will ring aswell because of the bridged alerting enabled. Station A is busy and doesnt want to answer the call, Station B pick-up the handset and automatically answers the bridged...

a little help please.

Hi Guys, Just want to ask if we can do bar users in doing a call-fwd on international numbers but local country (mobile, landline or ext-to-ext) are allowed. CM 5.2

where are you performing the ssh connection? On DOM-0 or Utilities domain? It should work already. The process that I'm doing is that, I do an ssh connection on U-DOM, which i will land on /home/admin directory. I will upload the files onto this folder, then do su - root/sroot01, once root is...

when you at /home/admin folder do an su - root/root01 then perform the command "chmod 777 *" then try to transfer the file to /vsp-template

For H.248 Gateways, do they have the same timeslot with G650? Or depends on the capacity of the gateway?